In many applications it is desirable to automatically control the level of an audio signal. Traditionally, such an audio signal processor is known as an automatic gain control (AGC), compressor, or limiter. A special type of such signal processors contains a (simple) estimator of loudness level, such that the processor attempts to control the level in a way that corresponds to the perceived loudness of the signal.
The limiters are a common type of audio signal processors. Such pre-existing limiters of audio signals were generally designed to limit signal transients or limit the maximum signal amplitude. Soft-clipping limiters achieve their limiting function in combination with a look-ahead delay, such that a saturating response can be applied—as opposed to a “hard clipping” limiter. However, in either case, such limiters generally operate at a time-scale ranging from single audio samples to a few ms duration. They control a technical property of the signal. Typically the goal of such limiters is for their processing to be “transparent”, i.e. near-inaudible. Hence, such limiters are neither suitable for, nor intended for, controlling nor limiting the loudness of the signal.
The auditory system has some properties of loudness perception that are roughly comparable to energy integration, with an order of magnitude of 20-100 ms.
Furthermore, speech (a common type of audio signal) requires a detector with an integration time of at least around 400 ms, in order for the measured level to be fairly steady for speech that is homogeneous (i.e. with constant loudness).
All dynamic range compressors/processors may be categorized as either feed-forward, meaning the side-chain and thus the level-control is based on a level measurement of the input signal. Or feed-back, meaning the side-chain is based on measuring the level of the output signal.
Some pre-existing compressors or AGCs can perform signal attenuation, based on an RMS level detector—which may be considered a primitive loudness level estimator. However, those processors that have been based on a feedback topology could only indirectly perform the “limiting” function. Because the side-chain is inherently a feedback loop, such a compressor will continually approximate the appropriate attenuation gain value. Hence, it may suffer from “over-shoots”, and/or may adapt too slowly to a change of the input signal requiring limiting.
Alternatively, some pre-existing compressors are based on a feed-forward topology. Typically, such processors have performed the (loudness) level estimate in their side-chain based on a short, fixed time-scale and subsequently applied smoothing/low-pass filtering to the control signal. Even though such a design may employ look-ahead delay, it cannot—in the general case—achieve limiting of the output loudness level. The calculation of each attenuation gain value, in pre-existing compressor, would be based on a single output value from its level detector. Furthermore, if a pre-existing dynamic range compressor were to perform a kind of limiting of the loudness level, it would require parameter settings of an infinitely high “ratio” and a very fast “attack”. Doing so might cause the processor to severely distort the signal.
Consequently, none of the existing audio processors are capable of combining the “limiting” property with the “loudness control” property. That is, processing the audio such that the loudness level, estimated on a given time-scale, is prevented from exceeding a specified loudness level threshold.
“Loudness limiter” processing has become increasingly relevant in the past decade. In broadcasts in radio/TV and other media, regulations may require that the programme itself—or commercials within/between programmes—must not exceed a certain loudness level, as measured on a specified time-scale. Both international and national standards and recommendations have been published in recent years, specifying and supporting such regulations, by organizations including ITU-R, EBU, ATSC, ARIB, and BCAP. Regulations of maximum loudness levels are also being specified in other areas, such as in the cinema, and for personal mobile audio devices.
The only ways to comply with such regulations have involved measuring, with a loudness meter, the loudness level (at the specified time-scale). If the programme or production in question was found to exceed the specification it would either need to be remixed in post-production, attenuating its loudest passages, and then measured again. That is a time-consuming process and requires an extended work-flow. Or the programme could simply be attenuated in its entirety, according to how much the measurement found necessary; however this is also undesirable, because of the overall loudness of the program would consequently also be attenuated—quite unnecessarily.